Capability negotiation in a telecommunications network

ABSTRACT

A method of negotiating a call capability between signalling points in a telecommunications system. The method comprises sending a capability preference or prioritised list of preferences from an originating signalling point to a terminating signalling point or signalling transfer point, at the Call Control level. A capability acceptance is returned from the terminating signalling point or signalling transfer point to the originating signalling point at the Call Control level, if the terminating signalling point or signalling transfer point accepts a preference sent by the originating signalling point.

This application is a divisional application of U.S. patent applicationSer. No. 09/573,500 filed on May 16, 2000 now U.S. Pat. No. 6,671,367,which claims priority from United Kingdom patent application number9911441.5 filed on May 17, 1999, United Kingdom patent. applicationnumber 9914654.0 filed on Jun. 22, 1999, United Kingdom patentapplication number 9914700.1 filed on Jun. 23, 1999, United Kingdompatent application number 9915366.0 filed on Jul. 2, 1999 and UnitedKingdom patent application number 9921647.5 filed on Sep. 15, 1999, thedisclosures of which are incorporated herein by reference.

FIELD OF THE INVENTION

The present invention relates to capability negotiation in atelecommunications network and in particular, though not necessarily, tothe negotiation of a suitable speech codec.

BACKGROUND TO THE INVENTION

Telecommunications networks currently rely to a large extent upon theSignalling System no.7 (SS7) as the mechanism for controlling callconnections and for handling the transfer of signalling informationbetween signalling points of the networks. Typically, one or moreapplication and user parts at a given signalling point will make use ofSS7 to communicate with peer application and user parts at some othersignalling point. Examples of user parts are ISUP (ISDN User Part) andTUP (Telephony User Part) whilst examples of application parts are INAP(Intelligent Network Application Part) and MAP (Mobile ApplicationPart). The conventional SS7 protocol stack includes Message TransferParts MTP1, MTP2, and MTP3 which handle the formatting of signallingmessages for transport over the physical layer as well as variousrouting functions.

There has been considerable interest of late amongst thetelecommunications community in using non-standard (i.e.non-conventional within the telecommunications industry) signallingtransport mechanisms in telecommunications networks in place of theconventional SS7 mechanisms. The reasons for this are related both toimprovements in efficiency as well as potential cost savings. Muchconsideration has been given for example to the use of Internet Protocol(IP) networks to transport signalling information between signallingpoints. IP networks have the advantage that they make efficient use oftransmission resources by using packet switching and are relatively lowin cost due to the widespread use of the technology (as opposed tospecialised telecommunication technology). There is also interest inusing other transport mechanisms including AAL1/2/5, FR etc.

The ISUP standard which deals with the setting-up and control of callconnections in a telecommunications network is closely linked to the SS7signalling transport mechanism and does not readily lend itself to usewith other non-standard transport technologies such as IP and AAL2. Assuch, several standardisation bodies including the ITU-T, ETSI, andANSI, are currently considering the specification of a signallingprotocol for the control of calls, which is independent of theunderlying transport mechanism. This can be viewed as separating outfrom the protocol, Bearer Control functions which relate merely toestablishing the parameters (including the start and end points) of the“pipe” via which user plane data is transported between nodes, and whichare specific to the transport mechanism. The new protocol, referred toas Transport Independent Call Control (TICC), retains Call Controlfunctions such as the services invoked for a call between given callingand called parties (e.g. call forwarding), and the overall routing ofuser plane data.

The new network architecture resulting from the separation of the calland Bearer Control levels results in an open interface appearing betweena Call Control entity and a Bearer Control entity, where these entitiesare referred to as a Media Gateway Controller and a Media Gatewayrespectively. The open interface is referred to hereinafter as X-CP,examples of which are the MEGACO work of the IETF and the H.248 work ofITU Study Group 16 (SG16).

Traditionally, fixed telephone networks make use of Pulse CodeModulation to transport user plane data, e.g. voice, facsimile, etc,between network nodes. Modern cellular networks on the other hand oftenuse one or more coders/decoders (referred to as “codecs”) to compressvoice signals for efficient transmission across the air interface andwithin the cellular networks themselves. Where a telephone callconnection extends between two networks (or terminals) which supportdifferent or multiple speech codecs, a negotiation may be carried outbetween the terminals to decide upon an appropriate codec. If thisnegotiation is not carried out, the result may be a requirement fortranscoding at the interface between the networks, i.e. conversion fromone form of speech coding to another. Transcoding is expensive in termsof resources, significantly degrades speech quality, and introduces aprocessing time delay. Codec negotiation is therefore the preferredoption.

In addition to codec negotiation, there is often a need in conventionaltelecommunications networks to negotiate other functionality andparameters. For example, it may be desirable to negotiate securitycapabilities such as voice ciphering and data encryption betweenterminals or nodes in telecommunications networks.

SUMMARY OF THE PRESENT INVENTION

According to a first aspect of the present invention there is provided amethod of negotiating a call capability between signalling points in atelecommunications system, the method comprising:

-   -   sending a capability preference or prioritised list of        preferences from an originating signalling point to a        terminating signalling point or signalling transfer point, at        the Call Control level; and    -   returning a capability acceptance from the terminating        signalling point or signalling transfer point to the originating        signalling point at the Call Control level, if the terminating        signalling point or signalling transfer point accepts a        preference sent by the originating signalling point.

It will be appreciated that in some cases, e.g. where the terminatingsignalling point or signalling transfer point does not accept acapability preference (or one of a list of preferences) sent by theoriginating signalling point, no acceptance message may be returned inwhich case a default capability is assumed by both points.Alternatively, a default message may be returned indicating that thedefault codec is to be used. If no codec can be agreed upon, then incertain situations a call may be released due to networkincompatibility.

The present invention is particularly suited to negotiating speech codeccapabilities between signalling transfer points located in differenttelecommunications networks. For example, in Japanese telecommunicationsnetworks, the invention may be used to negotiate the use of one ofVSELP, PSI-CELP, or μ-law coding, where μ-law coding is the defaultcoding. However, the invention is also applicable to negotiating othercapabilities including security capabilities (e.g. voice ciphering anddata encryption).

The protocol used to conduct the negotiation may be TICC, or may be aspecific protocol also employed at the CC level, i.e. a User PlaneCapability Negotiation protocol.

Where the Call Control and Bearer Control levels are controlled byseparate protocols, a signalling point reacts to the selection of acapability at the Call Control level by notifying the Bearer Controllevel, if the selection affects the bearer level. If appropriate,notifications may be subsequently sent at the bearer level betweenbearer switching points to enable the establishment of appropriatebearer level resources.

Preferably, the signalling point or signalling transfer point is a MediaGateway Controller. More preferably, the Media Gateway Controllercommunicates with one or more Media Gateways which exist at the BearerControl level.

Despite the fact that certain options supported for a capability at theBearer Control level may be known at the Call Control level, the CallControl level will not necessarily know the current availability ofthose options at the Bearer Control level. Preferably therefore, uponreceipt of a capability preference or prioritised list of preferences ata terminating signalling point or signalling transfer point, the CallControl level conducts a negotiation with the Bearer Control level todetermine option availability at the Bearer Control level. Morepreferably, this negotiation occurs between a Media Gateway Controllerof the Call Control level and a Media Gateway of the Bearer Controllevel.

Where a prioritised list of preferences is sent from an originatingsignalling point to a Media Gateway Controller, the Controllerpreferably modifies the list to remove preferences which it knows arenot supported by the associated Media Gateway. The Controller then sendsthe modified list to the Media Gateway which selects the preference withthe highest priority which the Media Gateway can support at that time.The Gateway may then reserve the resources necessary for that preferenceand advises the Media Gateway Controller of the preference. The MediaGateway Controller may then return a capability acceptance to theoriginating Media Gateway Controller.

According to a second aspect of the present invention there is provideda signalling point arranged to negotiate a call capability with anothersignalling point in a telecommunications system, the method comprising:

-   -   means for sending a capability preference or prioritised list of        preferences to a terminating signalling point or signalling        transfer point, at the Call Control level; and    -   means for receiving a capability acceptance from the terminating        signalling point or signalling transfer point at the Call        Control level, which capability acceptance is sent if the        terminating signalling point or signalling transfer point        accepts a preference sent by the originating signalling point.

According to a third aspect of the present invention there is provided aMedia Gateway Controller of a telecommunications system, the MediaGateway Controller comprising:

-   -   means for receiving a capability preference or prioritised list        of preferences from a peer Media Gateway Controller, where said        capability preference or prioritised list of preferences relate        to a connection to be set-up over the telecommunications system;    -   means for communicating with a Media Gateway associated with the        Media Gateway Controller to determine the availability of the        received preference(s) at the Media Gateway; and    -   means for returning a capability preference acceptance message        to said peer Media Gateway Controller in dependence upon the        determined availability at the Media Gateway.

According to a fourth aspect of the present invention there is provideda Media Gateway of a telecommunications system, the Gateway comprising:

-   -   means for receiving a capability preference or prioritised list        of preferences from a Media Gateway Controller, wherein said        capability preference or prioritised list of preferences relate        to a connection to be set-up over the telecommunications system;    -   means for selecting a preference on the basis of the        availability of the preferences at the Media Gateway; and    -   means for sending the selected preference to said Media Gateway        Controller.

According to a fifth aspect of the invention there is provided a methodof negotiating protocol options between first, second and third nodes ina telecommunications network using separated call control and bearercontrol protocols, the method comprising:

-   -   transmitting a first call control message from the first node to        the second node specifying protocol options supported by the        first node;    -   transmitting a second call control message from the second node        to the third node specifying protocol options supported by both        the first and second nodes; and    -   selecting a protocol option from the protocol options specified        in the second control message.

Said step of selecting a protocol option from the protocol optionsspecified in the second control message may be carried out by the thirdnode.

Each said call control message may include a preference level associatedwith each specified protocol option.

The first node may be an originating node. The third node may be aterminating node.

The method may further include a step of determining whether the bearerlevel between the first and second nodes is affected by said selectingstep, and if the bearer level is affected taking action to modify theparameters of the bearer level between the first and second nodes.

The method may also include determining whether the bearer level betweenthe second and third nodes is affected by said selecting step, and ifthe bearer level is affected taking action to modify the parameters ofthe bearer level between the second and third nodes.

According to a sixth aspect of the present invention there is provided amethod of setting-up a speech call connection in a telecommunicationssystem where the Call Control protocol is independent of the bearertransport mechanism, the method comprising:

-   -   negotiating a first speech codec between an originating        signalling point of the system and a first terminating        signalling point;    -   establishing a call connection over the transport mechanism        between the originating signalling point and said first        terminating signalling point in dependence upon said first        speech codec;    -   subsequently negotiating a second, different speech codec        between said first terminating signalling point and a second,        new terminating signalling point;    -   notifying the originating signalling point of the second speech        codec; and    -   establishing a call connection between the originating        signalling point and said second terminating signalling point        wherein said first originating signalling point acts as an        intermediate signalling point and wherein the first mentioned        call connection is modified if necessary to support said second        speech codec.

It will be appreciated that embodiments of the present invention enablethe smooth transfer of a call connection between different terminatingsignalling points, by modifying the connection between the originatingsignalling point and the original terminating signalling point toreflect the new codec. The final end-to-end connection is completed byestablishing a connection between the original terminating signallingpoint and the new terminating signalling point based upon the new codec.Preferably, this latter connection is established after the firstconnection is modified, although this need not be the case.

It will also be appreciated that the method of the present invention isonly applicable when the speech codec negotiation between the originalterminating signalling point and the final terminating signalling pointdoes not result in the first speech codec. If the result is the firstspeech codec, then there may be no need to modify the originalconnection.

Preferably, the step of notifying the originating signalling point ofthe second speech codec comprises sending an appropriate Call Control(CC) message from the original terminating signalling point to theoriginating signalling point. This CC message may be a Modify requestmessage.

Preferably, the Call Control protocol is a Transport Independent CallControl (TICC) protocol.

The present invention is applicable to the evolution of existingtelecommunication network such as mobile networks based on GSM, DAMPS,PDC, etc, as well as to future generation networks such as UMTS.

According to a seventh aspect of the present invention there is provideda telecommunications system in which the Call Control protocol isindependent of the bearer transport mechanism, the system comprising:

-   -   means for negotiating a first speech codec between an        originating signalling point of the system and a first        terminating signalling point;    -   means for establishing a call connection over the transport        mechanism between the originating signalling point and said        first terminating signalling point in dependence upon said first        speech codec;    -   means for subsequently negotiating a second, different speech        codec between said first terminating signalling point and a        second, new terminating signalling point;    -   means for notifying the originating signalling point of the        second speech codec; and    -   means for establishing a call connection between the originating        signalling point and said second terminating signalling point        wherein said first originating signalling point acts as an        intermediate signalling point and wherein the first mentioned        call connection is modified if necessary to support said second        speech codec.

According to an eighth aspect of the present invention there is provideda signalling point of a telecommunications network in which the CallControl protocol is independent of the bearer transport mechanism, thesignalling point comprising:

-   -   processing means for negotiating a first speech codec with a        first terminating signalling point;    -   means for establishing a call connection over the transport        mechanism between the originating signalling point and said        first terminating signalling point in dependence upon said first        speech codec;    -   means for receiving a notification of a second speech codec        which has been negotiated between said first terminating        signalling point and a second, new terminating signalling point;        and    -   means for modifying the first mentioned call connection if        necessary to support said second speech codec.

According to a ninth aspect of the present invention there is provided amethod of setting up a call connection between first and second mobiletelephone networks at least one of which comprises a Tandem FreeOperation (TFO) device located outside of the Radio Access Networkpart(s) of the mobile network and in which the Call Control protocol isindependent of the bearer transport mechanism, the method comprising thesteps of:

-   -   conducting a negotiation between said TFO device of one of the        mobile networks and a peer TFO device of the other mobile        network to determine a suitable speech codec; and    -   notifying said Radio Access Network of the determined speech        codec by sending a Call Control (CC) message from the associated        TFO device.

It may be the case that each of the mobile networks comprises a TandemFree Operation (TFO) device located outside of the Radio Access networkpart(s) of the mobile network. However, this need not be the case; andone of the mobile networks may have a TFO device within the radio AccessNetwork.

The two mobile telephone networks may be coupled to one another via aPSTN. Preferably, said negotiation is carried out using the CodecMismatch Resolution and Optimisation Procedure in the TFO protocol, withTFO messages being sent using in-band signalling.

Preferably, said the or each TFO device located outside of a RadioAccess network is located at a Gateway MSC (GMSC) which provides aninterface between the mobile network and foreign networks, e.g. a PSTN.

The present invention is particularly suited to setting up a callconnection between subscribers, one of whom is a subscriber of aUniversal Mobile Telecommunication System (UMTS) network. In this case,the Radio Access Network is preferably a UMTS Terrestrial Radio AccessNetwork (UTRAN), with the TFO device of the network being located on theedge of the UMTS core network.

According to an tenth aspect of the present invention there is providedapparatus for setting up a call connection between first and secondmobile telephone networks at least one of which comprises a Tandem FreeOperation (TFO) device located outside of the Radio Access Networkpart(s) of the mobile network and in which the Call Control protocol isindependent of the bearer transport mechanism, the apparatus comprising:

-   -   means for conducting a negotiation between said TFO device of        one of the mobile networks and a peer TFO device of the other        mobile network to determine a suitable speech codec; and    -   means for notifying said Radio Access Network of the determined        speech codec by sending a Call Control (CC) message from the        associated TFO device.

According to a eleventh aspect of the present invention there isprovided a method of setting up a call connection between first andsecond mobile telephone networks each of which comprises a Tandem FreeOperation (TFO) device located outside of the Radio Access Networkpart(s) of the mobile network and in which the Call Control protocol isindependent of the bearer transport mechanism, the mobile networks beingcoupled via a Public Switched Telephone Network (PSTN), the methodcomprising the step of:

-   -   conducting a negotiation between two mobile terminals        subscribing to the first and second mobile networks respectively        to determine a suitable speech codec, wherein the negotiation is        conducted using Call Control protocol signalling messages        exchanged between the Radio Access Network parts and the        respective TFO devices, and ISUP messages sent between the TFO        devices.

For example, a list of codecs which are available to an originatingmobile network may be sent from the TFO device of that network to thepeer TFO device using an ISUP Initial Address Message (IAM). The messagemay additionally include the codec preferred by the originating mobilenetwork. A subsequent ISUP message sent in the backward direction willindicate the codec type selected by the terminating mobile terminal. Inorder to enable this negotiation procedure, it may be necessary tomodify the ISUP standard.

According to a twelfth aspect of the present invention there is providedapparatus for setting up a call connection between first and secondmobile telephone networks each of which comprises a Tandem FreeOperation (TFO) device located outside of the Radio Access Networkpart(s) of the mobile network and in which the Call Control protocol isindependent of the bearer transport mechanism, the mobile networks beingcoupled via a Public Switched Telephone Network (PSTN), the apparatuscomprising:

-   -   means for conducting a negotiation between two mobile terminals        subscribing to the first and second mobile networks respectively        to determine a suitable speech codec, wherein the negotiation is        conducted using Call Control protocol signalling messages        exchanged between the Radio Access Network parts and the        respective TFO devices, and ISUP messages sent between the TFO        devices.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 illustrates a number of signalling points in a telecommunicationsnetwork; and

FIG. 2 (case A) illustrates signalling flows between the signallingpoints of FIG. 1 according to a first embodiment of the invention;

FIG. 2 (case B) illustrates signalling flows between the signallingpoints of FIG. 1 according to a second embodiment of the invention;

FIG. 3 illustrates a telecommunications network comprising Media GatewayControllers and Media Gateways;

FIG. 4 is a flow diagram illustrating a generic capability negotiationprocess used in the network of FIG. 3.

FIG. 5 illustrates signalling flows where an intermediate node isinvolved in the negotiation between an originating and a terminatingnode;

FIG. 6 illustrates a further set of signalling flows where anintermediate node is involved in the negotiation between an originatingand a terminating node;

FIG. 7 illustrates schematically a telecommunications system of knowndesign;

FIG. 8 illustrates schematically a telecommunications system;

FIG. 9 illustrates a first set of signals associated with codecnegotiation in the system of FIG. 8 and according to a first embodimentof the present invention; and

FIG. 10 illustrates a second set of signals associated with codecnegotiation in the system of FIG. 8 and according to a second embodimentof the present invention.

DETAILED DESCRIPTION OF CERTAIN EMBODIMENTS

There is illustrated in FIG. 1 a portion of a telecommunications networkcomprising two signalling points referred to hereinafter as Nodes A andC. These nodes may be for example telephone exchanges or switches andmay belong to the same network operator or to different networkoperators. In the present example, Node A is the originating node towhich a calling party (not shown) is connected whilst Node C representsa terminating node to which a called party is connected. Each of thesignalling points comprises a Call Control (CC) part and a BearerControl (BC) part, i.e. the Call Control and Bearer Controlfunctionalities are separated out into two distinct protocol layers. TheCC parts form a Call Control level which is responsible for performingfunctions such as call forwarding as well as other routing and controlfunctions. The BC parts are responsible for establishing anddimensioning pipes between BC parts for transporting user plane data.

Considering now the BC level, this comprises a bearer network which maybe for example an IP network. Within the IP network there are one ormore bearer switching points, although only one such point isillustrated in FIG. 1 (Node B). For the IP network, these bearerswitching points will be IP routers. It will be appreciated that wherethe bearer network is an ATM or AAL2 network the bearer switching pointswould be ATM or AAL2 switches respectively.

In the event that a calling party initiates a call, e.g. by taking histelephone off-hook, the originating signalling point Node A receives atthe CC level information from the source (i.e. caller) which defines thebearer resource requirements. The originating signalling pointdetermines, on the basis of the source information and/or thecapabilities of the signalling point and its home network, a list ofpossible options which must be negotiated with the terminatingsignalling point Node C.

Node A then signals the list of possible options in a message to the CCpart with which it wishes to negotiate a specific capability (node C inFIG. 1). The message indicates the level of preference for each option.

Node C uses the priority levels specified by Node A for each option inthe list to select the most preferred option that it supports, i.e. NodeC does not choose an option if it supports another option in the listfor which Node A has indicated a higher preference. Node C signals theselected option to Node A.

If the results of this negotiation affect the bearer connection level,appropriate actions are taken at the bearer level to adapt to theoutcome of the negotiation. This is done by the BC protocol. Two casesare possible: forward bearer modification (case A in FIG. 2) or backwardbearer modification (case B in FIG. 2).

An example of a capability which may be negotiated using the abovemethod is codec capability. Generally, speech is transcoded to PCM inthe radio access network because this is the only speech format allowedby traditional fixed telephone networks. However, as transcodingsignificantly deteriorates the speech quality, some cellular standards(e.g. GSM, PDC) specify methods to avoid transcoding when a connectionis established between two compatible terminals (e.g. two GSM terminalswith a common codec). Since most current mobile terminals supportseveral codecs, these methods also implement codec negotiation buthave-some important drawbacks compared to the generic negotiationmechanism presented here.

Speech codecs are either tightly coupled to mobile environments (in PDC,codec negotiation is a service of the Mobile Application Part (MAP)protocol) or do not optimise the use of hardware and transmissionresources (in GSM, codec negotiation is part of the Tandem FreeOperation (TFO) protocol). Two transcoders (TRAUs) and a 64 Kbps channelare allocated always for a speech connection independently of whethertranscoding actually happens or not). Currently, different mobilestandards deal with the problem in different ways and there haspreviously been no possibility to harmonize solutions.

Another example of a capability which may require negotiation issecurity capabilities (voice ciphering, data encryption, etc.). This iscurrently of great interest in public telecommunication networks andwill become more so in the near future. However, there are multiple waysof protecting user plane information against unwanted observers. Thereare multiple ciphering algorithms and data encryption algorithmscurrently deployed and new ones appearing continuously. Therefore, a wayto negotiate security capabilities in public telecommunication networkswill be required shortly and this is provided by the present invention.

The above description has referred generally to a BC level and a CClevel. FIG. 3 illustrates in more detail a telecommunications systemhaving split CC and BC levels. The CC level comprises a number of MediaGateway Controllers (MGC_A, MGC_B MGC_C) whilst the BC level comprises anumber of Media Gateways (MG_1 to MG_6). A first pair of Media Gateways(MG_1, MG_6) provide a bearer connection over an AAL2 network, whilst asecond pair of Media Gateways (MG_2, MG_5) provide a bearer connectionover an IP network. A third pair of Media Gateways (MG_3, MG_4) providea bearer connection over an STM network. As is apparent from FIG. 3,each of the Media Gateway Controllers is arranged to control either twoor three Media Gateways via an open interface (X-CP_1 to X-CP_3).

As has already been described, capability negotiation takes place at theCC level, i.e. between Media Gateway Controllers. In addition, the X-CPsprovide means for the Media Gateways to “inform” their-controlling MediaGateway Controllers of the capabilities and options which the MediaGateways support (e.g. capabilities=compressed speech, options=list ofcodecs). The X-CPs also provide a means whereby a Media Gateway can beinvolved in a capability negotiation, thus ensuring that a capabilityselection is based not only on the capabilities specified for thegateway but also on the current availability of those capabilities. Ifthis were not the case, there is a danger that a Media Gateway might nothave the resources to support an option negotiated on its behalf by aMedia Gateway Controller. Consider for example the case where a MediaGateway Controller needs to provide an Intelligent Network (IN) servicefor a call and which requires in-band announcements. If the MediaGateway Controller selects a codec by itself, based on a knowledge ofthe codecs supported by the Media Gateway, it is possible that the MediaGateway does not currently have available a TRAU for such a codec. TheMedia Gateway will therefore reject the connection, resulting in a calldrop.

To illustrate the solution to this problem, consider the situation wherea Media Gateway Controller receives from a peer Media Gateway Controllera connection set-up message containing a list of possible optionsrelating to a specific negotiable capability. The message indicates thelevel of preference for each option. The receiving Media GatewayController reads the received list and removes from the list thoseoptions which it knows are not supported by the Media Gateway which hasbeen selected for the connection. The Media Gateway Controller thensends the modified list of options to the Media Gateway over the X-CPinterface.

The receiving Media Gateway chooses the option having the highestpriority and which the Gateway is currently able to support. The MediaGateway then seizes the resources needed to support this option andcommunicates the option to the Media Gateway Controller over the X_CPinterface. Upon receipt of the selected option, the Media GatewayController signals the selection to the peer Media Gateway Controller asdescribed above.

The generic capability negotiation process described above is furtherillustrated by the flow diagram of FIG. 4.

In many countries operators are requested by local authorities toprovide interception of calls for legal purposes. In order to enablelegal interception, operators usually require full access to user planedata (i.e. the actual speech or other data carried by the so-called“pipe” at the bearer level) for each call inside their networks. Withthe introduction of TICC, user plane data may be transported over avariety of technologies (e.g. ATM, IP etc.) which allow for thetransport of encrypted user plane data. Therefore unless the LegalIntercept point can determine the security and codec (coding anddecoding) characteristics of the user plane data the call will not beable to be successfully intercepted. Therefore, the operator needs toknow the codecs and encryption (security) algorithms used inside each BCsection (between two BC nodes) of a call.

On the other hand, transcoding and translation of security algorithmssignificantly reduce the quality of service of a call and introduceextra delays. Therefore, it is desirable to reduce the number of (oravoid if possible) transcoding points and translation of securityalgorithm points required from end to end for each call.

A mechanism for solving this problem will now be described withreference to the sequence of operations shown in FIG. 5. The callcontrol part of Node A (see FIG. 1) transmits a message to the callcontrol part of Node B which includes a list of options supported byNode A, together with a preference level for each such option. Althoughthe mechanism is particularly suited for use in TICC it could also beused in an independent protocol specifically adapted for performingnegotiation, and this is indicated in FIG. 5 by the initials UPCN (UserPart Capability Negotiation).

The call control part of Node B then sends a message to the call controlpart of Node C, which includes a list of options supported by both NodeA and Node B, together with a preference level for each option. Node Cthen selects the supported option with the highest preference level, andreturns a message to Node B which includes the selected option. Node Bin turn sends a message to Node A specifying the selected option.

FIG. 5 then shows two possible sequences of operation, labelled case Aand case B. In Case A, if the bearer level is affected by thenegotiation at the call control level described above, Node A takesappropriate action at the bearer level. For example, it may be necessaryfor Node A to change the size of the pipe needed to support the selectedoption. A BC message is then sent from Node A to Node B specifyingwhatever parameters are needed. Node B then analyses the selected optionand takes appropriate action at the bearer level if the bearer level isaffected by the negotiation. A BC message is then sent from Node B toNode C specifying whatever parameters are needed for the bearer levelbetween nodes B and C.

In case B the sequence of operations at the bearer level starts at NodeC and ends at Node A. Node C analyses the selected option and takesappropriate action at the bearer level, if the bearer level if effectedby the negotiation. A BC message is then sent from Node C to Node Bspecifying whatever parameters are needed. Node B analyses the selectedoption and takes appropriate action at the bearer level if the bearerlevel is effected by the negotiation. A BC message is then sent fromNode B to Node A specifying whatever parameters are needed.

It should be understood that cases A and B are alternatives, but othercases are also possible. In this regard, it should be appreciated thatthe BC protocol between nodes A and B can be different from that betweennodes B and C, for example the protocols could be ATM and IPrespectively. It may be necessary to modify the parameters for thebearer level connection between one pair of nodes, but not between theother pair of nodes.

It should also be appreciated that the negotiation described above cantake place during call establishment or during the call itself. Thelatter may apply, for example, where a user wishes to start usingencryption during a call.

This procedure is able to reduce the number of points in the network atwhich the generic capability needs to be changed. In particular, themechanism specified above can be used to minimise the number oftranscoding points or translation of security algorithm points forend-to-end calls traversing multiple CC sections. The solution appliesfor calls traversing one or multiple networks.

The GCN mechanism includes having the initiating CC node include thelist of options with their preference level and the terminating nodeselect a supported option using the preference levels indicated by theoriginating node. The aforementioned negotiation mechanism is usefulwhen the negotiation is carried out between two CC nodes only. Theproposed adaptation expands the GCN mechanism to cases when more thantwo CC nodes intervene in the negotiation. That is, cases when the calltraverses multiple CC nodes belonging to one or more telecommunicationoperators. The adaptation involves the following sequence of operations:

1. The initiating CC node sends its list of supported options with thelevel of preference associated to each one.

2. Transit CC nodes analyse the received list of options, deleteunsupported options from the list and forward the list to the next node.

3. The terminating CC node: analyses the received list of options withtheir associated priorities and selects the supported option withhighest indicated priority.

In the case of legal interception it allows the Legal Interception point(i.e. the node at which interception is to take place) to receive thesecurity and coding characteristics of the user plane data to enablesuccessful interception of the call.

With the proposed new architecture of FIG. 1, where the Call Controlprotocol is independent of the transport mechanism, codec negotiationmust be performed by TICC for a call before a user plane transportconnection is established for this call. The reason is that the amountof transport resources required to support a call depends on the codecselected for this call (i.e. on the outcome of the codec negotiationprocedure). In some cases, a codec is initially selected for a call but,due to a change in circumstances, this codec must be changed for a newcodec in a later phase of the call. Two important situations where thisarises are the following:

1) Interactions with Intelligent Network (IN) services for the purposeof providing announcements and redirection of calls to a called partyeither automatically or based on input from the calling party. A firstcodec must be selected to provide the announcements. Later the call isforwarded/redirected to another party and a second codec is used.

2) Call Forwarding on No-Reply (CFNR) which is a so-called supplementaryservice. A first codec is negotiated between A and B parties followingwhich the call is forwarded to a C party because B fails to answer in apredefined time. A second codec is required based upon the requirementsof party C.

It is recognised that, in both of the cases 1) and 2) above, as well asin other related situations, a call connection must be established overthe transport mechanism in dependence upon the first selected codec, tocarry announcement, tones, etc. When the codec is subsequently changed,the call connection may need to be modified to support the new codec.

There will now be described with reference to FIG. 6 a signallingprocess for use in networks using the TICC protocol and which is capableof establishing an end-to-end telephone connection for a speech call,where the call originates from an originating signalling point Node Aand is initially directed to a first node Node B or signalling point.The call is subsequently routed to some other terminating signallingpoint Node C. The illustrated process relates specifically to theinvoking of an IN service, where a calling party is initially connectedto an IN network node which plays a pre-recorded message to the callingparty before routing the call to a final terminating signalling point,or to a CFNR service, where a call is forwarded to a new terminatingsignalling point after the called party, connected to the initialterminating signalling point, fails to answer the call.

The signalling sequence comprises the following sequential steps:

1. Node A establishes a call to node B. Codec X is selected for thiscall.

2. A transport connection with appropriate transport resources for theselected codec is set up between nodes A and B.

3. Node B forwards/redirects the call to node C. Node C does not supportcodec X and selects codec Y.

4. Node B requests Node A to modify the codec choice for this call fromcodec X to codec Y.

5. If needed, the transport connection between Node A and Node B ismodified to suit codec Y. FIG. 6 illustrates both the case where themodification to the transport connection is made in the forwarddirection and the alternative case where the modification is made in thebackward direction.

6. A transport connection suitable for codec Y is established betweenNodes B and C.

7. TICC completes the call establishment (TICC ACM+TICC ANM).

TICC is ISUP based, therefore the signalling message names are takenfrom ISUP. However, ISUP does not include codec negotiation or codecmodification procedures so a new pair of TICC messages (TICC modifyrequest/confirm) is needed to provide codec modification capabilities.

It will be appreciated that, whilst FIG. 6 illustrates only that themodification made to the Node A to Node B transport connection may bemade in either the forward or the backward direction, other transportconnection modifications may also be made in either the forward or thebackward direction. Also, the transport connections may be establishedin either the forward or backward direction.

The mechanism described may be employed to establish a speech callconnection between more than three signalling points or nodes. Forexample, one or more transit nodes may be present between theoriginating signalling point Node A and the original terminatingsignalling point Node B and/or between the original terminatingsignalling point Node B and the final terminating signalling point NodeC. It will also be appreciated that the call may be further transferredor relayed from Node C to yet another signalling point (i.e. a Node D).This would involve a negotiation to determine whether codec Y issuitable for Node D, and if not a possible modification to the transportconnection between Node A and Node B and between Node B and Node C. Thisprocess may be extended to any number of nodes.

Conventional Public Switched Telephone Networks (PSTN) digitally encodespeech data for transmission using Pulse Code Modulation (PCM). On theother hand, digital mobile telephone networks make use of more advancedcoding techniques such as CELP and Adaptive Multi-Rate (AMR) coding,which achieve higher compression ratios than can be achieved with PCM.In many mobile networks, coding and decoding of speech is carried out atthe mobile terminals themselves. Providing that a call is made betweentwo mobile terminals both registered with the same network it may bepossible to transmit encoded speech data from end to end.

In the event that a call is from a mobile terminal registered with amobile network, to a terminal which is a subscriber of a “foreign”network, end to end transmission of encoded speech data may not bepossible, depending upon the nature of the foreign network and of anyintermediate networks which connect the originating mobile network tothe foreign network (the same will of course be true where the calloriginates at the foreign network).

Consider the telecommunications system of FIG. 7 which illustrates twothird generation Universal Mobile Telecommunication System (UMTS)networks 1,2 which are coupled via a conventional PSTN/ISDN network 3.The UMTS networks 1,2 each comprise a UMTS Terrestrial Radio AccessNetwork (UTRAN) 4 having Radio Network Controllers (RNCs) 5 and BaseTransceiver Stations (BTSs) 6. A UTRAN 5 passes compressed speech databetween a mobile terminal (not shown) and a Mobile Switching Centre(MSC) 7 which routes incoming and outgoing call connections.

Assume that a call originates from a subscriber of one of the UMTSnetworks 1 and is made to a subscriber of the other of the UMTS networks2. The call is routed via the PSTN 3 using respective Transit nodes8—one of which is a Gateway MSCs (GMSCs)—of the UMTS networks 1,2. Ashas already been noted, the PSTN 3 uses PCM to encode speech data. Nowit is important that any speech data transferred through the PSTN 3 isin a form which can be understood by that network. This is necessary,for example, to allow the PSTN 3 to insert operator announcements into aspeech call, to perform voice prompting services, etc, as well as toallow the operator of the PSTN 3 to monitor call, e.g. for securitypurposes. It is therefore necessary to “transcode” speech data at theGMSCs 8 of the UMTS networks 1,2 prior to passing the data to the PSTN3, i.e. the speech data is converted from a mobile network speech codingformat to PCM. Similarly, PCM data received at the GMSCs 8 must beconverted to the appropriate mobile network speech coding format

Transcoding consumes considerable processing resources at a GMSC 8 andalso results in a perceivable degradation in speech quality. In order toat least partially overcome these disadvantages, Tandem Free Operation(TFO) devices may be introduced into the speech connection at the GMSCs8. Outgoing speech data continues to be converted to PCM, but the leastsignificant bits of each PCM sample are “stolen” by the TFO device. Thestolen bits form a channel which has sufficient bandwidth (i.e. 8Kbits/sec) to carry the original coded data. The TFO device at theterminating UMTS network reassembles the coded data for forwarding tothe associated UTRAN whilst the received PCM data is discarded (unlessit has been modified by the PSTN, e.g. by the addition of an operatorannouncement). In this way, TFO makes PCM data available to the PSTN 3,whilst still allowing the end to end transmission of efficiently codedspeech data.

In the event that intermediate devices within the PSTN/ISDN alter thePCM bit stream, the TFO devices detect the change and “fall-back” topassing the PCM coded speech between the TFO devices, i.e. they nolonger pass on the compressed voice data.

The speech codecs available to a mobile network depend upon the natureof the network and possibly upon the nature of a terminal using thenetwork. It will be apparent that the end to end use of a single codecis only possible when two networks are both capable of using the samecodec. Assuming that the GMSCs of two mobile communication networks areaware of the codec capabilities of the networks to which they belong, itis possible for them to negotiate and agree upon a common codec. Indeed,a suitable protocol is provided for in the ETSI recommendation GSM 08.62(version 7.0.0, release 1998).

A problem arises in attempting to implement TFO in mobile networks suchas are illustrated in FIG. 7, where the TFO devices are located on thefringe of the mobile networks, i.e. outside of the UTRAN (conventionallyTFO devices are located within the radio access networks). There iscurrently no mechanism for exchanging information, concerning codecsnegotiated between TFO devices, between TFO devices and radio accessnetworks, in such networks.

FIG. 8 illustrates a modified system architecture in which the GMSCs ofthe two UMTS networks have incorporated thereinto TFO devices 9, whilstFIG. 9 illustrates signalling associated with call set-up between twonodes in respective UMTS mobile telephone networks. Nodes 1 and 4represent MSCs, whilst the two middle Nodes 2 and 3 represent GMSCs.

Call set-up signalling within the UMTS networks is conducted at the CallControl (e.g. TICC) level, call set-up being initiated by an InitialAddress Message (IAM) sent from an MSC to the associated GMSC. This IAMuses a Generic Capabilities Negotiation (GCN) mechanism to determine anumber of parameters for the call connection. In particular, the IAMcontains a list of codecs supported by the originating UMTS network, aswell as the preferred codec. The originating side GMSC selects a codecfrom the transmitted list of codecs, and signals its selection back tothe MSC in the Message (Selected Codec 1). Subsequently, a callconnection is established at the bearer level (e.g. AAL2 or IP) withsufficient bandwidth to support the selected codec.

To enable end-to-end codec negotiation it is proposed to add the GCN tothe ISUP protocol. This end-to-end codec negotiation will maximize thepossibility of the end points utilizing the same codec type. If the endpoints use the same voice encoding algorithm, TFO is able to pass thecompressed voice through the PCM network without degrading the voicequality due to unnecessary transcoding. This also enables transmissionsavings by minimizing the bearer requirements within the originating andterminating networks that support compressed voice (e.g. AAL2 or IPbearer transport).

FIG. 9 illustrates the use of GCN enhanced ISUP messages to bridge thesignalling “gap” between the two UMTS networks. At the terminating UMTSnetwork, an IAM is sent using TICC from the GMSC to the MSC. The MSC inthis case accepts use of codec 1 and signals this back to the GMSC againusing TICC. Subsequently, the bearer level connection is established atthe terminating UMTS network. The PSTN relays the codec acceptance tothe originating side GMSC. As the originally proposed codec has beenaccepted, there is no need to change the bearer level connection at theoriginating UMTS network. However, if there is a change in the codec,this must be sent from the originating side GMSC to the MSC, so that thebearer level connection can be modified, e.g. to increase the bandwidthof the connection. In FIG. 9, ACM indicates an Address Complete Messageand ANM indicates an Answer Message.

FIG. 10 illustrates a second embodiment of the invention. This solutionrelies on the optional Codec Mismatch Resolution and Optimisationprocedure in the TFO protocol to detect incompatible codecs. When TFOdetects codec incompatibility, it can trigger a Codec modificationprocedure to modify the codec used by the terminals. TFO specifies therules for resolving codec mismatch (i.e. which codec to select). The TFOprotocol then triggers the TICC signalling to modify the codec used onthe call, to enable compatible codecs in the two end terminals. A changefrom the codec initially suggested by the originating UMTS network mayrequire a modification to the bearer level connection established ineither UMTS network.

The embodiments described above minimise unnecessary speech transcodingdue to intermediate PCM networks as well as allowing for the optimalallocation of user plane equipment (e.g. transcoding units) and/or userplane resources (e.g. bandwidth) to support the service level of aparticular call in public telecommunication networks.

Negotiation of the codec type will be required in existing networks andnew networks that support only PCM encoded voice in order to minimizeunnecessary transcoding in the speech path. The UMTS and GSMsubscriber's speech quality will not be degraded unnecessarily whentheir calls traverse existing PCM core networks having TFO support. Whenthe GCN mechanism is introduced into the TICC protocol, it is likely tobe carried in a transparent method (APM User).

It will be appreciated by the person of skill in the art that variousmodifications may be made to the above described embodiments withoutdeparting from the scope of the present invention.

1. A method of negotiating protocol options between first, second andthird nodes in a telecommunications network using separated call controland bearer control protocols, the method comprising the steps of:transmitting a first call control message from the first node to thesecond node specifying protocol options supported by the first node;transmitting a second call control message from the second node to thethird node specifying protocol options supported by both the first andsecond nodes; if there is a protocol option supported by both the firstand second nodes, selecting a protocol option from the protocol optionsspecified in the second control message, wherein said step of selectinga protocol option from the protocol options specified in the secondcontrol message is carried out by the third node; and if there is not aprotocol option supported by both the first and second nodes, selectinga default capability.
 2. A method according to claim 1, wherein theprotocol option is one selected from the group consisting of speechcodecs and security capabilities.
 3. A method according to claim 1,wherein each said call control message includes a preference levelassociated with each specified protocol option.
 4. A method according toclaim 1, wherein the first node is an originating node.
 5. A methodaccording to claim 1, wherein the third node is a terminating node.
 6. Amethod according to claim 1, which further includes determining whetherthe bearer level between the first and second nodes is affected by saidselecting step, and if the bearer level is affected taking action tomodify the parameters of the bearer level between the first and secondnodes.
 7. A method according to claim 1, which further includesdetermining whether the bearer level between the second and third nodesis affected by said selecting step, and if the bearer level is affectedtaking action to modify the parameters of the bearer level between thesecond and third nodes.